D-Link DPH-400G, DPH-400GE User Manual

Safety Notices

Please read the following safety notices before installing or using this phone.
They are crucial for the safe and reliable operation of the device.
 Please use the external power supply that is included in the package. Other

power supplies may cause damage to the phone, affect the behavior or
induce noise.

 Before using the external power supply in the package, please check with

home power voltage. Inaccurate power voltage may cause fire and damage.

 Please do not damage the power cord. If power cord or plug is impaired, do

not use it, it may cause fire or electric shock.

 The plug-socket combination must be accessible at all times because it

serves as the main disconnecting device.

 Do not drop, knock or shake it. Rough handling can break internal circuit

boards.

 Do not install the device in places where there is direct sunlight. Also do not

put the device on carpets or cushions. It may cause fire or breakdown.

 Avoid exposure the phone to high temperature, below 0℃ or high humidity.

Avoid wetting the unit with any liquid.

 Do not attempt to open it. Non-expert handling of the device could damage it.

Consult your authorized dealer for help, or else it may cause fire, electric
shock and breakdown.

 Do not use harsh chemicals, cleaning solvents, or strong detergents to clean

it. Wipe it with a soft cloth that has been slightly dampened in a mild soap
and water solution.

 When lightning, do not touch power plug or phone line, it may cause an

electric shock.

 Do not install this phone in an ill-ventilated place.
 You are in a situation that could cause bodily injury. Before you work on any

equipment, be aware of the hazards involved with electrical circuitry and be
familiar with standard practices for preventing accidents.

Table of Content

1 INTRODUCING VOIP PHONE ..................................................................................................... 6

1.1 SIMPLE INTRODUCTION ............................................................................................................... 6

1.2 DELIVERY CONTENT .................................................................................................................... 6

1.3 KEYPAD ....................................................................................................................................... 7

1.4 PORT FOR CONNECTING ............................................................................................................... 8

1.5 ICON INTRODUCTION ................................................................................................................... 9

1.6 LED INTRODUCTION .................................................................................................................. 10

2 INITIAL CONNECTING AND SETTING .................................................................................. 12

2.1 CONNECT THE PHONE ................................................................................................................ 12

2.1.1 Connect to network ............................................................................................................... 12

2.1.2 Power adaptor connection ..................................................................................................... 13

2.2 BASIC INITIALIZATION ............................................................................................................... 13

2.2.1 Network settings ................................................................................................................... 13

3 BASIC FUNCTION ........................................................................................................................ 15

3.1 MAKING A CALL ........................................................................................................................ 15

3.1.1 Call Device ........................................................................................................................... 15

3.1.2 Call Methods......................................................................................................................... 15

3.2 ANSWERING A CALL .................................................................................................................. 15

3.3 DND .......................................................................................................................................... 16

3.4 CALL FORWARD ........................................................................................................................ 16

3.5 CALL HOLD ............................................................................................................................... 16

3.6 CALL WAITING .......................................................................................................................... 16

3.7 MUTE ........................................................................................................................................ 16

3.8 CALL TRANSFER ........................................................................................................................ 17

3.9 3-WAY CONFERENCE CALL ......................................................................................................... 17

3.10 MULTIPLE-WAY CALL ................................................................................................................. 17

4 ADVANCED FUNCTION .............................................................................................................. 19

4.1 CALL PICKUP ............................................................................................................................. 19

4.2 JOIN CALL .................................................................................................................................. 19

4.3 REDIAL / UNREDIAL................................................................................................................... 19

4.4 CLICK TO DIAL ........................................................................................................................... 20

4.5 CALL BACK ................................................................................................................................ 20

4.6 AUTO ANSWER ........................................................................................................................... 20

4.7 HOTLINE .................................................................................................................................... 20

4.8 APPLICATION ............................................................................................................................. 20

4.8.1 SMS ...................................................................................................................................... 20

4.8.2 Memo .................................................................................................................................... 21

4.8.3 Voice Mail ............................................................................................................................ 21

4.9 PING .......................................................................................................................................... 21

4.10 PROGRAMMABLE KEY CONFIGURATION .................................................................................... 21

5 OTHER FUNCTIONS .................................................................................................................... 25

5.1 AUTO HANDDOWN .................................................................................................................... 25

5.2 BAN ANONYMOUS CALL............................................................................................................ 25

5.3 BAN OUTING ............................................................................................................................. 25

5.4 DIAL PLAN ................................................................................................................................. 25

5.5 DIAL PEER ................................................................................................................................. 25

5.6 AUTO REDIAL ............................................................................................................................ 26

5.7 CALL COMPLETION .................................................................................................................... 26

5.8 RING FROM HEADSET ................................................................................................................ 26

5.9 POWER LIGHT ............................................................................................................................ 26

5.10 HIDE DTMF .............................................................................................................................. 26

5.11 PASSWORD DIAL ................................................................................................ ........................ 26

5.12 PRE DIAL ................................................................................................................................ ... 27

5.13 ACTION URL & ACTIVE URI ..................................................................................................... 27

5.14 PUSH XML ................................................................................................................................ 27

6 BASIC SETTING ............................................................................................................................ 28

6.1 KEYBOAD .................................................................................................................................. 28

6.2 SCREEN SET ............................................................................................................................... 28

6.3 RINGER SET ............................................................................................................................... 28

6.4 VOICE VOLUME ......................................................................................................................... 28

6.5 TIME & DATE ............................................................................................................................ 28

6.6 GREETING WORD....................................................................................................................... 29

6.7 LANGUAGE SETTING .................................................................................................................. 29

7 ADVANCED SETTINGS ............................................................................................................... 30

7.1 ACCOUNT .................................................................................................................................. 30

7.2 NETWORK .................................................................................................................................. 30

7.3 SECURITY .................................................................................................................................. 30

7.4 MAINTENANCE .......................................................................................................................... 30

7.5 FACTORY RESET ......................................................................................................................... 30

8 WEB CONFIGURATION .............................................................................................................. 31

8.1 INTRODUCTION OF CONFIGURATION .......................................................................................... 31

8.1.1 Ways to configure ................................................................................................................. 31

8.1.2 Password Configuration........................................................................................................ 31

8.2 SETTING VIA WEB BROWSER ...................................................................................................... 31

8.3 CONFIGURATION VIA WEB........................................................................................................ 32

8.3.1 BASIC .................................................................................................................................. 32

8.3.2 Network ................................................................................................................................ 37

8.3.3 VOIP ..................................................................................................................................... 45

8.3.4 Phone .................................................................................................................................... 59

8.3.5 Function Key ........................................................................................................................ 76

8.3.6 Maintenance.......................................................................................................................... 79

8.3.7 Security ................................................................................................................................. 88

8.3.8 Logout................................................................................................................................... 93

9 APPENDIX ...................................................................................................................................... 94

9.1 SPECIFICATION .......................................................................................................................... 94

9.1.1 Hardware .............................................................................................................................. 94

9.1.2 Voice features ....................................................................................................................... 94

9.1.3 Network features ................................................................................................................... 95

9.1.4 Maintenance and management.............................................................................................. 96

1 Introducing VoIP Phone

1.1 Simple Introduction

Thank you for your purchasing DPH-400G/DPH-400GE.
DPH-400G/DPH-400GE is a full-feature telephone that provides voice
communication over the same data network that your computer uses. This phone
functions not only much like a traditional phone, allowing to place and receive
calls, and enjoy other features that traditional phone has, but also it owns many
data services features which you could not expect from a traditional telephone.
This guide will help you easily using the various features and services available
on your phone.

1.2 Delivery Content

Please check whether the delivery contains the following parts:

IP Phone are designed to look like conventional phones, the following photo
shows a broad overview of the IP Phone.

Item

Description

IP Phone

DPH-400G/DPH-400GE Phone with display and
keypad.

Power Adapter

Power supply for telephone.

Network Cable

Used to access network for the phone.

Handset

Make phone calls with the phone‟s basic functions.

Handset Cord

Connected with the handset and the phone.

Quick Installation Guide

Quick install the DPH-400G/DPH-400GE
guide.

CD

Containing manual and quick installation guide.

Warranty Safety Information

Warranty Safety Information for
DPH-400G/DPH-400GE .

1.3 Keypad

Key

Key name

Function Description

Navigation

Navigation keys assist users for operating.
In idle state they have special function.
You can configure through the web page according
to your patterns of use.

Phonebook

Access to phone book, check the record list and
add new records and revise the record. When check
the phone book record, press this key again will
return to idle mode.

Mute

Press this key in calling mode, you can hear the
other side, and the other side cannot hear you.

Hold

Temporarily hold the active call during the talking;
press the key again to unhold the call. You also can
press this key then input the third party‟s phone
number and end with the # key during calling; you
can make a call with the third party and hold the
previous calling.

Transfer

Use the key to realize blind transfer or attended
transfer.

Conference

Use this key to realize the three party call.

Volume -/+

Turn down or turn up the volume by pressing these
two keys.

Redial

1. In the hook off /hands-free mode, use the key
to dial the last call number.

2. In stand-by mode, it has a function to check the
Outgoing Call.

Hands-free

Make the phone into hands-free mode.

Indicator

light

If power on, the indicator is light.

Soft key 1/2/3/4

Keys combination, include functions such as
History/PBook /DND /Menu /Del /Redial /Send /
Quit/Answer/Divert/Reject/Hold/Transfer/Conf/Cl
ose and so on.

Call logs

View the Missed call, Incoming Call and dialed
Call.

Digital

keyboard

Inputting the phone number or DTMF.

DSS keys

Programmable keys to let you customize with
different functions. You can configure them in the
web page.

1.4 Port for connecting

Port

Port name

description

Power switch

Input: 5V AC, 1A.

WAN

10/100M Connect it to Network.

LAN

10/100M Connect it to PC.

Headset

Port type: RJ-9 connector.

Handset

Port type: RJ-9 connector.

External console
interface

Port type: RJ-11 direct connector.

1.5 Icon introduction

Icon

Description

Call out.

Call in.

Call hold.

Auto answer.

Call mute.

Contact.

DND(Do not Disturb).

In hand free mode.

In handset mode.

In headset mode.

SMS.

Missed call.

Call forward.

1.6 LED introduction

Table 1 Programmable key LEDs for BLF

LED Status

Description

Steady green

The object is in idle status.

Slow blinking red

The object is ringing.

Steady red

The object is active.

Fast blinking red

The object is failed.

Off

The object is failed/ No subscribe

Table 2 Programmable key LEDs for Presence

LED Status

Description

Steady green

The object is online.

Slow blinking red

The object is ringing.

Steady red

The object is active.

Fast blinking red

The object is failed.

Off

The object is failed/ No subscribe

Table 3Line key LEDs

LED Status

Description

Steady green

The account is active

Fast Blinking green

There is an incoming call to the account

Slow Blinking green

The call is on hold/ Registration is unsuccessful

Off

The line is unapplied or idle

Table 4 Programmable key LEDs for line

LED Status

Description

Steady green

The account is active

Fast Blinking green

There is an incoming call to the account

Slow Blinking green

The call is on hold

Slow Blinking red

Registration is unsuccessful

Off

The line is not unapplied or idle

Table 5 Programmable key LEDs for MWI

LED Status

Description

Blinking green

There are new voice mails

Off

There is no new voice mail

Table 6 Power Indication LED

LED Status

Description

Steady red

Power on.

Fast Blinking red

There is an incoming call.

Off

Power off.

2 Initial connecting and Setting

2.1 Connect the phone

2.1.1 Connect to network

Step 1: Connect the IP Phone to the corporate IP telephony network. Before you
connect the phone to the network, please check if your network can work
normally.
You can do this in one of two ways, depending on how your workspace is set up.
Direct network connection—by this method, you need at least one available
Ethernet port in your workspace. Use the Ethernet cable in the package to
connect WAN port on the back of your phone to the Ethernet port in your
workspace. Since this VoIP Phone has router functionality, whether you have a
broadband router or not, you can make direct network connect. The following
two figures are for your reference.

Shared network connection—Use this method if you have a single Ethernet port
in your workspace with your desktop computer already connected to it. First,
disconnect the Ethernet cable from the computer and attach it to the WAN port
on the back of your phone. Next, use the Ethernet cable in the package to
connect LAN port on the back of your phone to your desktop computer. Your IP
Phone now shares a network connection with your computer. The following
figure is for your reference.

Step 2: Connect the handset to the handset port by the handset cable in the
package.
Step 3: connect the power supply plug to the DC 5V adapter port on the back
of the phone. Use the power cable to connect the power supply to a standard
power outlet in your workspace.
If your LCD screen displays different information from the above, you need
refer to the next section “Initial setting” to set your network online mode.
If your VoIP phone registers into corporate IP telephony Server, your phone is
ready to use.

2.1.2 Power adaptor connection

Make sure that the power you use is comply with the parameters of power
adaptor.

1. Plug power adaptor to power socket.

2. Plug power adaptor‟s DC output to the DC5V port of

DPH-400G/DPH-400GE to start up.

3. There will be displayed black line and “initializing... wait logon...” on the
screen. After finishing startup, phone will show greeting, current date and time
and so forth.

4. If phone has registered to the server, you can place or answer calls.

2.2 Basic Initialization

DPH-400G/DPH-400GE is provided with a plenty of functions and parameters
for configuration. User needs some network and VoIP knowledge so that user
could understand the meanings of parameters. In order to make user use the
phone more easily and convenient, there are basic configurations introduced
which is mandatory to ensure phone calls.

2.2.1 Network settings

Make sure that network is connected already before setting network of phone.
DPH-400G/DPH-400GE uses DHCP to get WAN IP configurations, so phone
could access to network as long as there is DHCP server in it. If there is no

DHCP server available, phone has to be changed WAN network setting to Static
IP or PPPoE.

Setting PPPoE mode (for ADSL connection)

1. Get PPPoE account and password first.

2. Press Menu->Settings->Advanced Setting, then enter passwords, and choose
network ->WAN->Net Mode, enter and choose PPPoE through navigation keys
and press the Save key.

3. Press Quit, then choose PPPoE Set, press Enter.

4. The screen will show the current information. Press Del to delete it, then input
your PPPoE user and password and press Save.

5. Press Quit six times to return to the idle screen.

6. Check the status. If the screen shows “Negotiating…” it shows that the phone
is trying to access to the PPPoE Server; if it shows an IP address, then the phone
has already get IP with PPPoE.

Setting Static IP mode (static ADSL/Cable, or no PPPoE / DHCP network)

1. Prepare the network‟s parameters first, such as IP Address, Net mask,
Default Gateway and DNS server IP address. If you don‟t know this information,
please contact the service provider or technician of network.

2. Press Menu->Settings->Advanced Setting, then enter passwords, and choose
network ->WAN->Net Mode, enter and choose Static through navigation keys
and press the Save key.

3. Press Quit, then choose Static Set, press Enter.

4. The screen will show the current information, and then press Del to delete.
Input your IP address, Mask, Gateway, DNS and press Save to save what you
input.

5. Press Quit six times to return to the idle screen.

6. Check the status, the screen shows “Static” .the screen shows the IP address
and gateway which were set just now, if the phone could display the right time, it
shows that Static IP mode takes effect.

Setting DHCP mode

1. Press Menu->Settings->Advanced Setting, then enter passwords, and choose

network ->WAN->Net Mode, enter and choose DHCP through navigation keys
and press the Save key.

2. Press Quit six times to return to the idle screen.

3. Check the status, the screen shows “DHCP”，If the screen shows the IP
address and gateway which were set just now, it shows that DHCP mode takes

effect.

3 Basic function

3.1 Making a call

3.1.1 Call Device

You can make a phone call via the following devices:

1. Pick up the handset, icon will be showed in the idle screen.

2. Press the Speaker button, icon will be showed in the idle screen.

3. Press the Headset button if the headset is connected to the Headset Port in
advance. The icon will be showed in the idle screen.

You can also dial the number first, and then choose the method you will use to
speak to the other party.

3.1.2 Call Methods

You can press an available line button if there is more than one account, then

1. Dial the number you want to call.

2. Press History softkey, use the navigation buttons to highlight your choice

(press Left/Right button to choose Missed Calls, Incoming Calls and
Outgoing Calls.

3. Press the RD button to call the last number called.

4. Press the programmable keys which are set as speed dial button.
Then press the Send button or Send softkey to make the call if necessary.

3.2 Answering a call

Answering an incoming call

1. If there is no other calling, you could choose the handle or press the speaker

button or use softkey-answer or press the headset to accept the call.

2. If you are on another call, press the fluctuation navigation key to answer the

new call.
During the conversation, you can alternate between Headset, Handset and
Speaker phone by pressing the corresponding buttons or picking up the handset.

3.3 DND

Press the DND softkey to active DND Mode. New incoming calls will be rejected and
the display will show: icon. Press the DND softkey to choose deactivate DND
mode. Incoming calls will be stored in the Call History.

3.4 Call Forward

This feature allows you to forward an incoming call to another phone number.
The display showed icon.
The following call forwarding events can be configured:

Off: Call forwarding is deactivated by default.
Always: Incoming calls are immediately forwarded.
Busy: Incoming calls are immediately forwarded when the phone is busy.
No Answer: Incoming calls are forwarded when the phone is not answered

after a specific period.
To configure Call Forward via Phone interface:

1. Press Menu ->Features->Enter->Call Forward->Enter.

2. There are 4 options: Off, Always, Busy, No Answer.

3. If you choose one of them (except Off), enter the phone number you want to
forward your calls to. Press Save to save the changes.

3.5 Call Hold

1. Press the Hold button or Hold softkey to put your active call on hold.

2. If there is only one call on hold, press the hold softkey to retrieve the call.

3. If there are more than one call on hold, press the line button, and the
Up/Down button to highlight the call, then press the Unhold button to retrieve
the call.

3.6 Call Waiting

1. Press Menu ->Features->Enter->Call Waiting->Enter.

2. Use the navigation keys to active or inactive call waiting.

3. Then press the Save to save the changes.

3.7 Mute

Press Mute button during the conversation, icon will be showed in the LCD.
Then the called will not hear you, but you can hear the called. Press it again to

get the phone to normal conversation.

3.8 Call transfer

1. Blind Transfer
During talk, press the key Transf, and then dial the nu mber that you want to
transfer to, and finished by "#". Phone will transfer the current call to the third
party. After finishing transfer, the call you talk to will be hanged up. User cannot
select SIP line when phone transfers call.

2. Attended Transfer
During talk, press the key Transf, then input the number that you want to transfer
to and press Send. After that third party answers, then press Transfer to complete
the transfer. (You need enable call waiting and call transfer first). If there are two
calls, you can just talk to one, and keep hold to the other one. The one who is
keep hold cannot speak to you or hear from you. In other way, if user wants to
invite the third party during the call, they can press Conf to make calls mode in
conference mode. If user wants to stop conference, user can press Split. (User
must enable call waiting and three way call first).
Note: the server that user uses must support RFC3515 or it might not be used

3. Semi-attended Transfer
During the talk, press Transf firstly, and then press Send after inputting the
number that you want to transfer. You are waiting for connection, now, press
Transf and the transfer will be done. (To use this feature, you need enable call
waiting and call transfer first).

3.9 3-way conference call

1. Press the Conf softkey during an active call.

2. The first call is placed on hold. Then you will hear a dial tone. Dial the
number to conference in, then press Send key.

3. When the call is answered, press Conf and add the first call to the
conference.

4. If you want to release the conference, press Split key.

Note: the server that user uses must support RFC3515 or it might not be used

(User must enable call waiting and three way call first).

3.10 Multiple-way call

In this phone you can registe 5 SIP account numbers and the 5 accuonts can be
used at the same time.There are four keys used as SIP line toleranted to make
calls in SIP accounts.It will blink when the account registed failed.
In order to convenience the enterprise the phone support multiple call answering ,

call hold and multi-line call.The user can answer 10 incoming call phones at
most,you can choose any call through pressing the fluctuation navigation key in
taiking and the other 9 calls will be in held.You also can press the fluctuation
navigation key to change the call and recover the talking then last call will be
held automatic.You also can define the four line keys as multi-line keys ,then
each line key will relate to a call and you can choose the talking through pressing
the line keys and recover the talking and the light to the line key will bright all
the time when in taking ,then the light of the call in held is sparking.
If user has 5 line calls and wants to invite the five party during the call, they can
press Conf or Transf “New Call”, press OK, enter the number ,then press Send
and wait for the other party to answer. When the multiple-way calls, you can
press the arrow keys to select a call.

4 Advanced function

4.1 Call pickup

Call pickup is implemented by simulating pickup function of PBX. it‟s that,
when A calls B, B rings but no answer, at this moment, C can hook off and input
an appointed prefix plus B‟s number, pick up A‟s call and talk with A.
The following chart shows how to configure an appointed pref ix in dial peer to
have call pick up function.

*1* means appointed prefix code. After making the above configuration, C can
dial *1* plus B‟s phone number to pick up A‟s call. User can set prefix in
random, in the case of no affecting current dialing rules.

4.2 Join call

When B is calling C, A can join in the existing call by inputting an appointed
prefix numbers plus B or C number, if B or C also supports join call.
The following chart shows how to configure an appointed pref ix in dial peer to
have join call function.

*2* means appointed prefix code. After making the above configuration, A can
dial *2* plus B or C number to join B and C‟s call. User can set prefix in random,
in the case of no affecting current dialing rules.

4.3 Redial / Unredial

If B is in busy line when A calls B, A will get notice: busy, please hang up. If A
want to connect B as soon as B is in idle, he can use redial function at the
moment and he can dials an appointed prefix number plus B‟s number to realize
redial function.
What is redial function? A can‟t not build a call with B when B is in busy, then
A will subscribe B‟s calling mode at 60 second intervals. Once B is available, A
will get reminder of rings to hook off, while A hooks off, A will call B
automatically. If at this time A is occupied temporarily and unwilling to contact
B, A also can cancel the redial function by dialing an appointed prefix plus B‟s

number before making the redial function.

*3* is appointed prefix code. After making the above configuration, A can dial
*3* plus B‟s phone number to make the redial function.
*4* is appointed prefix code. After configuration, A can dial *4* to cancel redial
function.
User can set prefix in random, in the case of no affecting current dialing rules.

4.4 Click to dial

When user A browses in an appointed Web page, user A can click to call user B
via a link (this link to user B), then user A‟s phone will ring, after A hooks off,
the phone will dial to B.

4.5 Call back

This function allows you dial out the last phone call you received.

4.6 Auto answer

When there is an incoming call, after no answer time, the phone will answer the
call automatically.

4.7 Hotline

You can set hotline number for every sip, and then enter the dialer interface and
after Warm Line Time, the phone will call out the hotline number automatically.

4.8 Application

4.8.1 SMS

1) Press Menu ->Application->Enter->SMS->Enter.

2) Use the navigation keys to highlight the options. You can read the message in
the Inbox/Outbox.

3) After view the new message, you can press Reply to reply the message, and
use the 123 softkey to change the Input Method, when enter the reply message,
press OK, then use the navigation keys to select the line from which you want to

send, then Send.

4) If you want to write a message, you can press New and enter message. Use the
123 softkey to change the Input Method. When you input the message you want
to send, press OK, then use the navigation keys to select the line from which you
want to send, then Send.

5) If you want to delete the message, after view the message, press Del, then you
have three options to choose: Yes, All, No.

4.8.2 Memo

You can add some memos to record some important things to remind you.
Press Menu->Application->Memo->Enter->Add.
There are some options to configure: Mode, Date, Time, text, Ring. When the
configuration is completed, press Save.

4.8.3 Voice Mail

1) Press Menu->Voice Mail->Enter.

2) Use the navigation keys to highlight the line for which you want to set, press
Edit, and use the navigation key to turn on the mode, and the input the number.
Press 123 softkey to choose the proper input method.

3) Press Save to save the change.

4) To view the new voicemail, Press the Voicemail softkey directly. Press Dial,
then you may be prompted to enter the password, then you can listen to your
new and old messages.

4.9 Ping

1) Press Menu-> Application->ping>Enter.

2) Input the IP you want ,and press start key ,if input wrong, you can press

“delete” to modification the IP.

3) After input the IP, wait a moment it will display”confirmation” , it meas ping

successful ,or means ping failed.

4.10 Programmable Key Configuration

The phone has 8 programmable keys which are able to set up to many functions
per key. The following list shows the functions you can set on the programmable
keys and provides a description for each function. The default configuration for
each key is N/A which means the key hasn‟t been set for any functions.

1. Set the type as Memory Key
Press Menu->Settings->Basic Setting->Enter->DSS Key, you have two options:

Line As DSS Keys and Memory As DSS Keys, choose one you want to make
the assignment, use the navigation key to choose the type as memory key. In the
Dial field, you have some options, such as Normal, Speed Dial, Intercom, BLF,
Presence, and MWI.

Speed dial

You can configure the key as a simplified speed dial key. This key function
allows you to easily access your most dialed numbers.

Intercom

You can configure the key for Push to talk code and it is useful in an office
environment as a quick access to connect to the operator or the secretary.

BLF

BLF is also called “Busy lamp field”, and it is used to prompt the user to pay
attention to the state of the object than has been subscribed, and used to
cooperate with the server to pick up the phone call. You can configure the key
for Busy Lamp Field (BLF) which allows you to monitor the status (idle, ringing,
or busy) of other SIP account. User can dial out on a BLF configured key. Please
refer to “LED Instruction” for more detail about the LED status in different
situation.
Note: In the Web interface, you can also set the pickup number to active the
pickup function. For example, if you set the BLF number as 212, and the pickup
number is 189, then when there is an incoming call to 212, press the BLF key, it
will call out the 189 automatically to pick up the incoming call on 212.

Presence

Presence is called present, and compared to the BLF, it can also check whether
object online.
Note: You can subscribe the BLF and presence station of the same number at the
same time.

MWI

When the key is configured as MWI, you are allowed to access voicemail
quickly by pressing this key.

CALL PARK

You need setting a server number, when you have set what represent Call park.
If you have a calling and you busy now, you could press the key and hear a
number, then you could choose other phone and input this number. so you can
directly recover call..

Call forward

When there is an incoming call, press the key and the incoming call can be
transferred to the number set up.

2. Set the type as Line
You can set these keys as line keys, and press it, it will enter dialer interface.

3. Set the type as Key Event
You can set these keys as Key Event, and the subtype have many options.
Choose one and it will have corresponding function.
 None

 MWI
 DND (Do Not Disable)
 Hold
 Transfer
 Phone Book
 Redial
 Pick up
 Join
 Auto Redial On
 Auto Redial Off
 Call Forwarding
 History
 Flash
 Memo
 Headset
 Release: Press the key you can end the call.
 Lock: Press the key you can lock the keyboard.
 SMS
 Call Back
 Power Light
 Hide DTMF
 Prefix
 Hot Desking: Pressing the key, you can clear all sip information and register

yourself sip information

 Agent

4. Set the type as Dtmf
You can configure the key as Dtmf. This key function allows you to easily dial
or edit dial number.

5. Set the type as Remote
You need to match a XML Phonebook address, pressing the button you can
directly access the corresponding remote phonebook.

6. Set the type as BLF List Key

It needs the cooperation with the Broadsoft server. The traditional BLF is
that every number will need to be subscribed, so if the numbers that
subscribed is so many that it will cause to obstruction. However, BLF List
Key will put the numbers that needed to be subscribed in a group, and the
phone use the URL of the group to subscribe and analyze the specific
information of each number such as number, name, state and so on according
to the notifications from the server. Then set the idle Memory key as BLF
List Key, later if the state of an object changes, the corresponding LED will
change.

7. set the type as Multicast

Set the multicast address and audio code, press this key to initiate the
multicast.

Notice: Detailed feature see 8.3.4.7

5 Other functions

5.1 Auto Handdown

1. Press Menu ->Features-> Enter->Auto Handdown-> Enter.

2. Set the Mode Enable through the navigation key, then set Time, unit is minute,
then press Save.

3. When the call ends, after the time that you have set, the phone will back to the
idle interface.

5.2 Ban Anonymous Call

1. Press Menu ->Features-> Enter->Ban Anonymous Call-> Enter.

2. Choose which sip you want to enable Ban Anonymous Call, and then press
Enter, choose Enabled or Disabled through navigation key.

3. If you choose Enabled, the others can‟t call the phone by anonymous. If you
choose Disabled, the others can call the phone by anonymous.

5.3 Ban Outing

1 . Press Menu ->Features-> Enter->ban outgoing> Enter

2.Enable the function, then you can not call any number.

5.4 Dial Plan

1. Press Menu ->Features-> Enter->Dial Plan-> Enter.

2. The following plans you can set: Press # to Send, Timeout to Send, Timeout,
Fixed Length Number, Press # to Do BXFER, BXFER On Onhook, AXFER On
Onhook. You can enable or disable each dial plan.

5.5 Dial Peer

1. Press Menu ->Features-> Enter->Dial Peer-> Enter.

2. Press Add to enter the Edit interface, and then input some information. For
example: Number: 1T, Dest.: 0.0.0.0, Port: 5060, Mode: SIP, Alisa: all:3333,
Suffix: no suffix, Del Len: 0. Then press Save. Then press Save.

3. Input 1+number (1234) in the dial interface, you can dial out 3333.
You can refer to 8.3.3.4 DIAL PEER.

5.6 Auto Redial

1. Press Menu ->Features-> Enter->Auto Redial-> Enter.

2. Choose Mode Enabled or Disabled through the navigation key. If you choose
Enable, you also need to set Interval and Times, and then press Save.

3. After enable auto redial, calling out someone, if he is in busy, it will pop up a
prompt box whether to auto redial, press OK, the phone will call out him
according the Interval and Times that you set.

5.7 Call completion

1. Press Menu ->Features-> Enter->Call Completion-> Enter.

2. Enable the function through the navigation key, and then Save .

3. Call out others, if he is in busy, it will pop up a prompt Call Completion
Waiting number? Press OK, when he is in idle, it will pop up a prompt Call
Completion Call number? Press OK, the phone will call out the number
automatically.

5.8 Ring From Headset

1. Press Menu ->Features-> Enter->Ring From Headset-> Enter.

2. Enable this function through the navigation key, the phone connects the

headset, when the phone has an incoming call, it will ring from the headset.

5.9 Power Light

1. Press Menu ->Features-> Enter->Power Light-> Enter.

2. Enable this function through the navigation key.

5.10 Hide DTMF

1. Press Menu ->Features-> Enter->Hide DTMF-> Enter.

2. Through the navigation key to choose: Disabled, All, Delay, Last Show. When
you set up a call with others and need to input the DTMF, the DTMF will show
as you have set.

5.11 Password Dial

1. Press Menu ->Features-> Enter->Password Dial-> Enter.

2. Enable this function, you can also set Prefix and Length. For example，you

want call out 1234567 and you set Password Dial Prefix 123 and Password
Length 3, then enter the dial interface and input 1234567, and then the screen
will show 123***7.

5.12 Pre Dial

1. Press Menu ->Features-> Enter->Pre Dial-> Enter.

2. Through navigation key to enable the feature,and to realize the Pre Dial
function.

5.13 Action URL & Active URI

1. Action URL: The action that the phone carries out e.g. open dnd can produces
one URL, then the phone can send the HTTP Get of the URL to PC, then the
phone can report the action to the PC.

2. Active URI: Enter the web page of the phone, PHONE->FEATURE, input
Active URL Limit IP, You can input internet server (e.g. PC'IP), PC can send one
URL to the phone, the phone will produce one action for example open dnd, so
PC can control the phone.

5.14 Push XML

Enter the web page of the phone->PHONE->FEATURE, input Push XML
Server(e.g. PC‟IP), then PC can push text, SMS, phonebook, advertisement,,
execute etc. to phone to update the message or the phone makes an action.

6 Basic setting

6.1 Keyboad

1. Press Menu ->Settings-> Enter->Basic Setting-> Enter->Keyboard->Enter.

2. There are four items: DSS Keys, Multiplex, Long Click, SoftKey, You can set
up respectively on them. Press the key Enter to the interface, then use the
navigation keys to choose the function for the key according to you want.

3. Press the key OK to save.

6.2 Screen Set

1. Press Menu ->Settings-> Enter->Basic Setting-> Enter->Screen Set->Enter.

2. You can set Contrast and Brightness, press Enter and use the navigation keys
to set, then press the key Save.

6.3 Ringer Set

1. Press Menu ->Settings-> Enter->Basic Setting-> Enter->Ringer Set->Enter.

2. You can set Ringer Volume and Ringer Type, press Enter and use the
navigation keys to set, then press the key Save. In the Ringer Type, the default
system rings have nine and the custom ringtones have five that can be set
through the web page.

6.4 Voice Volume

1. Press Menu ->Settings-> Enter->Basic Setting-> Enter->Voice
Volume->Enter.

2. Use the navigation keys to turn down or turn up the voice volume, the press
the key Save.

6.5 Time & Date

1. Press Menu ->Settings->Enter->Basic Setting-> Enter->Time & Date->Enter.

2. You have two options to choose: Auto and Manual, use the navigation keys to
choose, then press Save.

6.6 Greeting Word

1. Press Menu ->Settings-> Enter->Basic Setting-> Enter->Greeting
Word->Enter.

2. You can enter the message and press Save, it will display in the phone screen
when the phone start up.

6.7 Language Setting

1. Press Menu ->Settings-> Enter->Basic Setting-> Enter->Language->Enter.

2. DPH-400G/DPH-400GE support three languages, you can use the navigation
keys to choose. The default two languages are English and Chinese.

7 Advanced settings

7.1 Account

Press Menu->Enter->Advanced settings, and then input the password to enter the
interface, the default password is 123. You can set it through the web page. Then
choose Account then press Enter, you can do some SIP settings.

7.2 Network

Press Menu->Enter->Advanced settings, and then input the password to enter the
interface. Then choose Network and press Enter, you can do network settings,
you can refer to 2.2.1 Network settings.

7.3 Security

Press Menu->Enter->Advanced settings, and then input the password to enter the
interface. Then choose Security, you can configure Menu Password, Keylock
Password, Keylock Status.

7.4 Maintenance

Press Menu->Enter->Advanced settings, and then input the password to enter the
interface. Then choose Maintenance and press Enter, you can configure Auto
Provision, TR069, Backup, and Upgrade.

7.5 Factory Reset

Press Menu->Enter->Advanced settings, and then input the password to enter the
interface. Then choose Factory Reset and press Enter, you can choose Yes or No.

8 Web configuration

8.1 Introduction of configuration

8.1.1 Ways to configure

DPH-400G/DPH-400GE has three different ways to different users.

 Use phone keypad.
 Use web browser (recommendatory way).
 Use telnet with CLI command.

8.1.2 Password Configuration

There are two levels to access to phone: root level and general level. User with
root level can browse and set all configuration parameters, while user with
general level can set all configuration parameters except SIP (1~5) ‟s that some
parameters cannot be changed, such as server address and port. User will has
different access level with different username and password.

 Default user with general level：

 username：guest
 password：guest

 Default user with root level：

 username：admin
 password：admin

The default password of phone screen menu is 123.

8.2 Setting via web browser

When this phone and PC are connected to network, enter the IP address of the
wan port in this phone as the URL (e.g. http://xxx.xxx.xxx.xxx/ or
http://xxx.xxx.xxx.xxx:xxxx/).
If you do not know the IP address, you can look it up on the phone‟s display by
checking menu-Status.
The login page is as below picture

8.3 Configuration via WEB

8.3.1 BASIC

8.3.1.1 Status

Status

Field name

Explanation

WAN

Shows the configuration information on WAN port,
including the connect mode of WAN port (Static,
DHCP, PPPoE), MAC address, the IP address of WAN
port, IP Gateway,ON or OFF bridge mode.

Accounts

Shows the phone numbers provided by the SIP LINE
1~5 servers.

The last line shows the version number.

8.3.1.2 Wizard

Wizard

Please select the proper network mode according to the network condition.
DPH-400G/DPH-400GE provide three different network settings:
 Static: If your ISP server provides you the static IP address, please select

this mode, and then finish Static Mode setting. If you don‟t know about
parameters of Static Mode setting, please ask your ISP for them.

 DHCP: In this mode, you will get the information from the DHCP server

automatically; need not to input this information artificially.

 PPPoE: In this mode, you must input your ADSL account and password.
You can also refer to2.2.1 Network setting to speed setting your network.

Choose Static IP MODE，click【NEXT】can configure the network and

SIP(default SIP1)simply, also can browse too. Click【BACK】can return to
the last page.

IP Address

Input the IP address distributed to you.

Subnet Mask

Input the Netmask distributed to you.

IP Gateway

Input the Gateway address distributed to you.

DNS Domain

Set DNS domain postfix. When the domain which you
input cannot be parsed, phone will automatically add
this domain to the end of the domain which you input
before and parse it again.

Primary DNS

Input your primary DNS server address.

Secondary DNS

Input your secondary DNS server address.

Display Name

Set the display name.

Server Address

Input your SIP server address.

Server Port

Set your SIP server port.

Authentication User

Input your SIP registered account name.

Authentication

Input your SIP registered password.

Password

SIP User

Input the phone number assigned by your VOIP
service provider.

Enable Registration

Start to register or not by selecting it or not.

Display detailed information that you manual configure.
Choose DHCP MODE，click【NEXT】can configure SIP(default

SIP1)simply, also can browse too. Click【BACK】can return to the last page.
Like Static IP MODE.
Choose PPPoE MODE，click【NEXT】can configure the PPPoE
account/password and SIP(default SIP1)simply, also can browse too. Click
【BACK】can return to the last page. Like Static IP MODE.

Service Name

It will be provided by ISP.

User

Input your ADSL account.

Password

Input your ADSL password.

Notice: Click【Finish】button after finished your setting, IP Phone will save
the setting automatically and reboot, After reboot, you can dial by the SIP

account.

8.3.1.3 Call Log

You can query all the dialed calls through this page.

Call Log

Field name

explanation

Start Time

Display the start time of the dialed calls.

Duration

Display the conversation time of the dialed calls.

Dialed Calls

Display the account /line of the dialed calls.

8.3.1.4 LANGUAGE

LANGUAGE

Field name

explanation

Language

Set the language of phone, English is default.

Greeting Words

The greeting message will display on LCD when
phone is idle. It can support 16 chars. the default chars
are VOIP PHONE.

Notice: the maximal length of the greeting message is twelve English
characters and five Chinese characters.

8.3.2 Network

8.3.2.1 WAN

WAN Status

Active IP Address

The current IP address of the phone.

Current Subnet
Mask

The current Netmask address.
MAC Address

The current MAC address of the phone.

Current IP Gateway

The current Gateway IP address.

MAC Timestamp

Shows the time of getting MAC address

Please select the proper network mode according to the network condition.
DPH-400G/DPH-400GE provide three different network settings:
 Static: If your ISP server provides you the static IP address, please select

this mode, and then finish Static Mode setting. If you don‟t know about
parameters of Static Mode setting, please ask your ISP for them.

 DHCP: In this mode, you will get the information from the DHCP server

automatically; need not to input this information artificially.

 PPPoE: In this mode, you must input your ADSL account and password.
You can also refer to 2.2.1 Network setting to speed setting your network.

Obtain DNS server
automatically

Select it to use DHCP mode to get DNS address, if
you don‟t select it, you will use static DNS server. The
default is selecting it.

Enable Vendor
Identifier

Enable/Disabled Vendor Identifier
Vendor Identifier

Custom vendor identification

If you use static mode, you need set it.

IP Address

Input the IP address distributed to you.

Subnet Mask

Input the Netmask distributed to you.

IP Gateway

Input the Gateway address distributed to you.

DNS Domain

Set DNS domain postfix. When the domain which
you input cannot be parsed, phone will automatically
add this domain to the end of the domain which you
input before and parse it again.

Primary DNS

Input your primary DNS server address.

Secondary DNS

Input your secondary DNS server address.

If you uses PPPoE mode， you need to make the above setting.
Service Name

It will be provided by ISP.

User

Input your ADSL account.

Password

Input your ADSL password.

Notice:

1) Click “Apply” button after finished your setting, IP Phone will save the
setting automatically and new setting will take effect.

2) If you modify the IP address, the web will not response by the old IP
address. Your need input new IP address in the address column to logon in
the phone.

3) If networks ID which is DHCP server distributed is same as network ID
which is used by LAN of system, system will use the DHCP IP to set WAN,
and modify LAN‟s networks ID (for example, system will change LAN IP
from 192.168.10.1 to 192.168.11.1) when system uses DHCP client to get IP
in startup; If system uses DHCP client to get IP in running status and

network ID is also same as LAN‟s, system will refuse to accept the IP to

configure WAN. So WAN‟s active IP will be 0.0.0.0.

8.3.2.2 Qos&VLAN

The VOIP phone support 802.1Q/P protocol and DiffServ configuration. VLAN
functionality can use different VLAN IDs by setting signal/voice VLAN and
data VLAN. The VLAN application of this phone is very flexible.

In chart 1, there is a layer 2 that switches without setting VLAN. Any broadcast
frame will be transmitted to the other ports except the send port. For example, a
broadcast information is sent out from port 1 then transmitted to port 2,3and 4.
In chart 2, red and blue indicate two different VLANs in the switch, and port 1
and port 2 belong to red VLAN, port 3 and port 4 belong to blue VLAN. If a
broadcast frame is sent out from port 1, switch will transmit it to port 2, the other
port in the red VLAN and not transmit it to port3 and port 4 in blue VLAN. By
this means, VLAN divide the broadcast domain via restricting the range of
broadcast frame transition.
Note: chart 2 use red and blue to identify the different VLAN, but in practice,
VLAN uses different VLAN IDs to identify.

QoS &VLAN

LLDP Settings

Enable LLDP

Enable LLDP by selecting it.

Enable Learning
Function

After enabling LLDP Learn, telephone can
automatically learn the data of DSCP, 802.1p, VLAN
ID from the switch. If the data is different from the
data of the LLDP server, telephone will change its
own value as the value of the switch (Synchronous
with VLAN in switch).

Package Interval

The time interval of sending LLDP Packet.

QoS Setting

Enable DSCP

Enable Dscp by selecting it.

SIP DSCP

Specify the value of the SIP Dscp.

Audio RTP DSCP

Specify the value of the Audio RTP Dscp.

WAN Port VLAN
Setting

Enable WAN Port
VLAN

Enable WAN Port VLAN by selecting it.

WAN Port VLAN
ID

Specify the value of the WAN Port VLAN ID, the
range of the value is 0-4095.

SIP 8021.p Priority

Specify the value of the signal 802.1p priority, the
range of the value is 0-7.

Audio 802.1p
Priority

Specify the value of the voice 802.1p priority, the
range of the value is 0-7.

LAN Port VLAN
Setting

LAN Port VLAN
Mode

Follow WAN: Follow the WAN ID.
Disable: Disable Port VALN.
Enable: Enable Port VLAN and specify the Port
VLAN ID different from WAN ID.

LAN Port VLAN
ID

Specify the value of the Port VLAN ID different from
WAN ID, the range of the value is 0-4095.

8.3.2.3 Service Port

You can set the port of telnet/HTTP/RTP by this page.

SERVICE PORT

Field name

explanation

Service Port
Settings

Web Server Type

Specify Web Server Type with HTTP or HTTPS.

HTTP Port
Set web browser port, the default is 80 port，if you

want to enhance system safety，you'd better change it

into non-80 standard port；
Example: The IP address is 192.168.1.70. and the port

value is 8090, the accessing address is
http://192.168.1.70:8090.

HTTPS Port

Before using the https, you must download https
authentication certification into the phone, then

set web browser port, the default is 443 port，if you

want to enhance system safety，you'd better change it
into non-443 standard port. You can access to the web

in https after rebooting the phone.

Telnet Port

Set Telnet Port, the default is 23. You can change the
value into others.
Example: The IP address is 192.168.1.70. the telnet
port value is 8023, the accessing address is telnet

192.168.1.70 8023.

RTP Port Range
Start

Set the RTP Start Port. It is dynamic allocation.

RTP Port Number

Set the maximum quantity of RTP Port, the default is

200.

Notice:

1) You need save the configuration and reboot the phone after set this page.

2) Please REBOOT the system if you modify the HTTP or telnet port
number (the new number should be greater than 1024.)

3) If you set 0 for the HTTP port, it will disable HTTP service.

8.3.2.4 TIME&DATE

Setting time zone and SNTP (Simple Network Time Protocol) server according
to your location, you can also manually adjust date and time in this web page.

TIME&DATE

Field name

explanation

Simple Network
Time Protocol
(SNTP) Settings

Enable SNTP

Enable SNTP by selecting it.

Enable DHCP Time

Enable DHCP Time by selecting it, then the
phone will automatically synchronize the standard
time.

Primary Server

Set SNTP Primary Server IP address.

Secondary Server

Set SNTP Secondary Server IP address.

Time Zone

Select the Time zone according to your location.

Resync Period

Set the time out, the default is 60 seconds.

12 -Hours Clock

Switch the time mechanism between 12 hours and 24
hours.

Default is 24 hours mode.

Date format

Specify the date display format.

Daylight Saving
Time Settings

Enable

Enable daylight saving time.

Offset(minutes)

Setup the variety length.

Month

Setup start and end month.

Week

Setup start and end week.

Day

Setup start and end day.

Hour

Setup start and end hours.

Minute

Setup start and end minutes.

Manual Time Settings

Notice: You need specify the above all items.

8.3.3 VOIP

8.3.3.1 SIP

Set your SIP server in the following interface.

SIP Config

Field name

explanation

SIP Line

Choose line to set info about SIP, there are 5 lines to choose. You can switch
by【Load】button.

Basic Settings

Status

Shows if the phone has been registered the SIP
server or not; or so, show Unapplied.

Server Address

Input your SIP server address.

Server Port

Set your SIP server port.

Authentication User

Input your SIP register account name.

Authentication
Password

Input your SIP register password.

SIP User

Input the phone number assigned by your VoIP
service provider. Phone will not register if there is
no phone number configured.

Display Name

Set the display name.

Proxy Server Address

Set proxy server IP address(Usually, Register SIP
Server configuration is the same as Proxy SIP
Server. But if your VoIP service provider give
different configurations between Register SIP Server
and Proxy SIP Server, you need make different
settings).

Proxy Server Port

Set your Proxy SIP server port.

Proxy User

Input your Proxy SIP server account.

Proxy Password

Input your Proxy SIP server password.

Domain Realm

Set the sip domain if needed, otherwise this VoIP
phone will use the Register server address as sip
domain automatically. (Usually it is same with
registered server and proxy server IP address).

Backup Server
Address

Input the Backup Server Address, if the primary
server is unavailable , then the phone will enable the
Backup Server Address

Backup Server Port

Specify the Backup Server Port

Server Name

Named the server

Enable Registration

Start to register or not by selecting it or not.

Codecs Settings

Disable
Codecs/Enable
Codecs

Use the navigation keys to highlight the desired one
in the Enable/Disable Codecs list, and press the
desired to move to the other list.

Advanced SIP
Setting

Forward Type

Select call forward mode, the default is Off
Off：Close down calling forward

Busy：If the phone is busy, incoming calls will be
forwarded to the appointed phone.
No answer： If there is no answer, incoming calls
will be forwarded to the appointed phone after a

specific.
Always：Incoming calls will be forwarded to the
appoint phone immediately.

The phone will prompt the incoming while doing
forward.

Forward Number

Specify the number you want to forward.

No Answer Forward
Wait Time

Specify the No Answer Forward Delay Time，if the
Forward Type is No answer, incoming calls will be

forwarded after the no answer forward wait time

Transfer Timeout

For the phone supports the transfer of certain special
features server, set interval time between sending
“bye” and hanging up after the phone transfers a
call.

Enable Hot Line

Specify Hot Line by selecting it

Hot Line Number

Specify Hot Line Number, the phone dial the hot
line number automatically at hands-free mode or
handset mode after warm line time

Warm Line Wait
Time

Specify the Warm Line Time

BLF Server

Ordinary BLF application is that the phone send
subscription package to the registered server, if your
server does not support subscription package, please
input the BLF server so that it can separate register
server and BLF server.

SIP Encryption

Enable/Disable SIP Encryption.

SIP Encryption Key

Set the key for sip encryption.

RTP Encryption

Enable/Disable RTP encryption.

RTP Encryption Key

Set the key for RTP encryption.

Enable Auto Answer

Enable Auto Answer by selecting it

Auto Answer
Timeout

Specify Auto Answer Time, the phone auto answers
the incoming call after Auto Answer Time

Enable Session Timer

Set Enable/Disable Session Timer, whether support
RFC4028.It will refresh the SIP sessions.

Session Timeout

Set the session timeout

Subscribe for MWI

Enable the Subscribe for MWI by selecting it, the
phone will send subscribe message for MWI to the
SIP Server

MWI Number

Specify the MWI Number; Please contact your
system administrator for the connecting code.
Different systems have different codes.

Subscribe Period(s)

Overtime of resending subscribe packet. Suggest
using the default configuration.

Conference Type

Specify the Conference Type, if you select the local,
you needn‟t input the conference number

Conference Number

Specify the network conference number, please
contact your system administrator for the network
conference number

Registration Expire(s)

Set expire time of SIP server register, default is 60
seconds. If the register time of the server requested
is longer or shorter than the expired time set, the
phone will change automatically the time into the
time recommended by the server, and register again.

Enable Service Code

If you want to realize the following function by the
server, please enter the On Code and Off Code
option, then when you choose to enable/disable
following function on your IP phone, it will send
message to the server, and the server will turn on/off
the function immediately.

DND On Code

Set the DND On Code, When you press the DND
hot key, the phone will send a message to the server,
and the server will turn on the DND function. Then
any calls to the extension will be rejected by the
server automatically. And the incoming call record
will not be displayed in the Call History.

DND Off Code

Set the DND Off Code, When you press the DND
hot key, the phone will send a message to the server,
and the server will turn off the DND function.

Always CFwd On
Code

Set the Always CFwd On Code, when you choose to
enable the always forward function on your phone, it

will send message to the server, and the server will
turn on the function immediately. When there are
calls to the extension, the server will always forward
it to the set number automatically. And the IP phone
will not show the record in the call history anymore.

Always CFwd Off
Code

Set the Always CFwd Off Code, when you choose to
disable the always forward function on your phone,
it will send message to the server, and the server will
turn off the function immediately.

Busy CFwd On Code

Set the Busy CFwd On Code, when you choose to
enable the busy forward function v on your phone, it
will send message to the server, and the server will
turn on the function immediately. When there are
calls to the extension, the server will forward it to
the set number automatically based the forward
type. And the IP phone will not show the record in
the call history anymore.

Busy CFwd Off Code

Set the Busy CFwd Off Code, when you choose to
disable the busy forward function on your phone, it
will send message to the server, and the server will
turn off the function immediately.

No Answer CFwd On
Code

Set the No Answer CFwd On Code, when you
choose to enable the on answer forward function on
your phone, it will send message to the server, and
the server will turn on the function immediately.
When there are calls to the extension, the server will
forward it to the set number automatically based the
forward type. And the IP phone will not show the
record in the call history anymore.

No Answer CFwd Off
Code

Set the No Answer CFwd Off Code, when you
choose to disable the busy forward function on your
phone, it will send message to the server, and the
server will turn off the function immediately.

Ban Anonymous On
Code

Set the Ban Anonymous On Code, When you
choose to enable the ban anonymous call function
on your IP phone, it will send information to the
server, and the server will enable the ban
anonymous call function for your IP phone
automatically.

Ban Anonymous Off
Code

Set the Ban Anonymous Off Code, When you
choose to disable the ban anonymous call function
on your IP phone, it will send information to the
server, and the server will disable the ban
anonymous call function for your IP phone

automatically.

Keep Alive Type

Specify the keep alive type, if the type is option, the
phone will send option sip message to server every
NAT Keep Alive Period(s), then the server responses
with 200 to keep alive. If the type is UDP, the phone
will send UDP message to server to keep alive every
NAT Keep Alive Period(s).

Keep Alive Interval

Set examining interval of the server, default is 60
seconds

User Agent

Set the user agent if have, the default is VoIP Phone

1.0

DTMF Type

Select DTMF sent mode, there are three modes:

 DTMF_In-band
 DTMF_RFC2833
 DTMF_SIP_INFO
 DTMF_AUTO

Different VoIP Service providers may provide
different modes.

DTMF SIP INFO
Mode

There are two options: send 10/11 and send * / #
Ring Type

Set ring type of each line

Enable Rport

Enable/Disable system to support RFC3581. Via
rport is special way to realize SIP NAT.

Enable PRACK

Enable or disable SIP PRACK function, suggest use
the default config.

Enable Long Contact

Set more parameters in contact field; connection
with SEM server

Convert URI

Convert # to %23 when send the URI.

Dial Without
Registered

Set call out by proxy without registration;
Ban Anonymous Call

Set to ban Anonymous incoming Call;

Enable DNS SRV

Support DNS looking up with _sip. udp mode

Enable Missed Call
Log

Enable the missed call log by it, the phone will save
the missed call log into the call history record and
display the missed calls on the idle screen, or won‟t
save the missed call log into the call history record
and display the missed calls on the idle screen.

Enable BLF List

Enable BLF List by selecting it, BLF list is a
function which can monitor the group status, it is not
one to one monitoring, but the information
feedback from the sever to decide which
BLF list will monitor

BLF List Number

Specify the BLF List Number

Respond 182 when

when there is a call in call waiting, the phone will

Call waiting

return 182 code

Server Type

Select the special type of server which is encrypted,
or has some unique requirements or call flows.

RFC Protocol Edition

Select SIP protocol version to adapt for the SIP
server which uses the same version as you select.
For example, if the server is CISCO5300, you need
to change to RFC2543; else phone may not cancel
call normally. System uses RFC3261 as default.

Local Port

Set sip port of each line

RFC Protocol Edition

Set Anonymous call out safely; Support
RFC3323and RFC3325;

Keep Authentication

Enable/Disable Keep Authentication System will
take the last authentication field which is passed the
authentication by server to the request packet. It will
decrease the server‟s repeat authorization work, if it
is enable.

Answer With A
Single Codec

Enable/Disable the function when call is incoming,
phone replies SIP message with just one codec
which phone supports.

Auto TCP

Set to use automatically TCP protocol to guarantee
usability of transport as message is above 1300 byte

Enable Strict Proxy

Support the special SIP server-when phone receives
the packets sent from server， phone will use the
source IP address, not the address in via field.

Enable GRUU

Set to support GRUU

Enable Display name
Quote

Set to make quotation mark to display name as the
phone sends out signal, in order to be compatible
with server.

Enable user=phone

Enable user=phone by selecting it, it is contained in
the invite sip message, in order to be compatible
with server

Click to talk

Set click to Talk (need practical software support).

Transport Protocol

Set transport protocols, TCP or UDP or TLS;

Use VPN

Phone use vpn ip to communicate

Enable DND

When the type of DND feature is line, enable the
DND feature for a line

SIP Global Settings

Strict Branch

Enable the Strict Branch, the value of the branch
must be in the beginning of z9hG4k in via field of
the invite sip message received, or the phone won‟t
response to the invite sip message.
Notice: the deployment will become effective in all
sip lines

Enable Group

Enable Group by selecting it, then the phone enable
the sip group backup function
Notice: the deployment will become effective in all
sip lines

Registration Failure
Retry Time

Specify the registration failure retry time, if the
phone register failed, the phone will register again
after registration failure retry time.
Notice: the deployment will become effective in all
sip lines.

8.3.3.2 STUN

In this web page, you can configure SIP STUN.
STUN: By STUN server, the phone in private network could know the type of
NAT and the NAT mapping IP and port of SIP. The phone might register itself to
SIP server with global IP and port to realize the device both calling and being
called in private network.

STUN

Field name

explanation

Simple Traversal of
UDP through NATs
(STUN) Settings

STUN NAT Traversal

Shows STUN NAT Transverse estimation, true
means STUN can penetrate NAT, while False
means not.

Server Address

Set your SIP STUN Server IP address.

Server Port

Set your SIP STUN Server Port.

Blinding Period

Set STUN blinding period(s). If NAT server finds
that a NAT mapping is idle after time out, it will
release the mapping and the system need send a
STUN packet to keep the mapping effective and
alive.

SIP Waiting Time

Specify the sip wait stun time; you can input the
time depended on your network condition.

Sip Line Using STUN

Choose line to set info about SIP, There are 4 lines to choose. You can switch
by 【Load】 button.

Use STUN

Enable/Disable SIP STUN.

Notice: SIP STUN is used to realize SIP penetration to NAT. If your phone
configures STUN Server IP and Port (default is 3478), and enable SIP Stun,
you can use the ordinary SIP Server to realize penetration to NAT.

8.3.3.3 DIAL PEER

This functionality offers you more flexible dial rule, you can refer to the
following content to know how to use this dial rule. When you want to dial an IP
address, the entry of IP addresses is very cumbersome, but by this functionality,
you can set number 156 to replace 192.168.1.119 here.

When you want to dial a long distance call to Beijing, you need dial an area code
010 before local phone number, but you can also dial number 1 instead of 010
after we make a setting according to this dial rule. For example, you want to dial
01062213123, but you need dial only 162213123 to realize your long distance
call after you make this setting.

To save the memory and avoid abundant input of user, add the follow functions:

1.* Match any single digit that is dialed.
If user makes the above configuration, after user dials 11 digit numbers started
with 13, the phone will send out 0 plus the dialed numbers automatically.

2. [] Specifies a range that will match digit. It may be a range, a list of ranges
separated by commas, or a list of digits.
If user makes the above configuration, after user dials 11 digit numbers started
with from 135 to 139, the phone will send out 0 plus the dialed numbers
automatically.
Use this phone you can realize dialing out via different lines without switch in
web interface.

DIAL PEER

Field name

explanation

Phone number

There are two types of matching conditions: one is full
matching, the other is prefix matching. In the Full
matching, you need input your desired phone number
in this blank, and then you need dial the phone number
to realize calling to what the phone number is mapped.
In the prefix matching, you need input your desired
prefix number and T; then dial the prefix and a phone
number to realize calling to what your prefix number
is mapped. The prefix number supports at most 30
digits.

Destination

Set Destination address. This is optional config item.
If you want to set peer to peer call, please input
destination IP address or domain name. If you want to
use this dial rule on SIP2 line, you need input

255.255.255.255 or 0.0.0.2 in it.SIP3 into 0.0.0.3.

Port

Set the Signal port, the default is 5060 for SIP.

Alias

Set alias. This is optional config item. If you don‟t set

Alias, it will show no alias.

Note: There are four types of aliases.

1) Add: xxx, it means that you need dial xxx in front of phone number, which
will reduce dialing number length.

2) All: xxx, it means that xxx will replace some phone number.

3) Del: It means that phone will delete the number with length appointed.

4) Rep: It means that phone will replace the number with length and number

appointed.
You can refer to the following examples of different alias application to
know more how to use different aliases and this dial rule.

Call Mode

Select different signal protocol, SIP

Suffix

Set suffix, this is optional config item. It will show no
suffix if you don‟t set it.

Delete Length

Set delete length. This is optional config item. For
example: if the delete length is 3, the phone will delete
the first 3 digits then send out the rest digits. You can
refer to examples of different alias application to know
how to set delete length.

Examples of different alias application

Set by web

explanation

example

You need set phone
number, Destination,
Alias and Delete Length.
Phone number is XXXT;
Destination is

255.255.255.255 (0.0.0.2)

and Alias is del.
This means any phone
No. that starts with your
set phone number will be
sent via SIP2 line after the
first several digits of your
dialed phone number are
deleted according to
delete length.

If you dial
“93333”, the
SIP2 server will
receive “3333”

This setting will realize
speed dial function, after
you dialing the numeric

key “2”, the number after

all will be sent out.

When you dial

“2”, the SIP1

server will
receive
33334444

The phone will
automatically send out
alias number adding your
dialed number, if your
dialed number starts with
your set phone number.

When you dial

“8309“, the SIP1

server will
receive
“07558309”

You need set Phone
Number, Alias and Delete
Length. Phone number is
XXXT and Alias is rep:
xxx
If your dialed phone
number starts with your
set phone number, the
first digits same as your
set phone number will be
replaced by the alias
number specified and
New phone number will
be send out.

When you dial

“0106228”, the

SIP1 server will
receive
“86106228”

If your dialed phone
number starts with your
set phone number. The
phone will send out your
dialed phone number
adding suffix number.

When you dial

“147”, the SIP1

server will
receive
“1470011”

8.3.4 Phone

8.3.4.1 Audio

In this page, you can configure voice codec, input/output volume and so on.

DSP Configuration

Field name

explanation

First Codec

The first preferential DSP
codec:G.711A/U,G.722,G.723.1, G.726-32,
G.729AB,ILBC,AMR,AMR-WB

Second Codec

The second preferential DSP codec:
G.711A/U,G.722,G.723.1, G.726-32,
G.729AB,ILBC,AMR,AMR-WB,NONE

Third Codec

The third preferential DSP codec:
G.711A/U,G.722,G.723.1, G.726-32 , G.729AB,
ILBC,AMR,AMR-WB,NONE

Fourth Codec

The forth preferential DSP codec:
G.711A/U,G.722,G.723.1, G.726-32 , G.729AB,
ILBC,AMR,AMR-WB,NONE

Fifth Codec

The fifth preferential DSP codec:
G.711A/U,G.722,G.723.1, G.726-32, G.729AB,

ILBC,AMR,AMR-WB,NONE

Sixth codec

The sixth preferential DSP codec:
G.711A/U,G.722,G.723.1,
G.726-32,G.729AB,ILBC,AMR,AMR-WB,NONE

Seventh Codec

The seventh preferential DSP codec:
G.711A/U,G.722,G.723.1, G.726-32,
G.729AB,ILBC,AMR,AMR-WB,NONE

Eighth Codec

The eighth preferential DSP codec:
G.711A/U,G.722,G.723.1, G.726-32, G.729AB,
ILBC,AMR,AMR-WB,NONE

Ninth Codec

The ninth preferential DSP codec:
G.711A/U,G.722,G.723.1, G.726-32, G.729AB,
ILBC,AMR,AMR-WB,NONE

Onhook Time

Specify the least reflection time of Hand down, the
default is 200ms.

Default Ring Type

Select Ring Type

Handset Volume

Specify Handset Volume grade.

Speakerphone
volume

Specify Speakerphone Volume grade.
Headset Volume

Specify Headset Volume grade.

Headset Ring
Volume

Specify Headset Ring Volume grade

Speakerphone Ring
Volume

Specify Speakerphone Ring Volume grade
ILBC Payload Type

Set ILBC payload type

ILBC Payload
Length

Set ILBC Payload Length
AMR Payload Type

Set AMR payload type

AMR-WB Payload
Type

Set AMR-WB payload type

G729AB Payload
Length

Set G729 Payload Length
G723.1 Bit Rate

5.3kb/s or 6.3kb/s is available

DTMF Payload
Type

Set DTMF Payload Type.

EnableVAD

Select it or not to enable or disable VAD. If enable
VAD, G729 Payload length could not be set over
20ms.

Enable MWI Tone

the phone will play MWI tone when a new MWI
comes

8.3.4.2 FEATURE

In this web page, you can configure Call Transfer, Call Waiting, 3 Ways Call,
Black List, white list Limit List and so on.

FEATURE

Field name

explanation

Do Not
Disturb

There are there options:
Disabled: The phone accept any normal incoming call
Phone: The phone rejects any incoming call, the caller will
automatically prompt hang up, but outgoing calls will not be
affected
Line: A line enabled DND will reject it's any incoming call

Ban
Outgoing

If you select Ban Outgoing to enable it, and you cannot dial out
any number.

Enable Call
Transfer

Enable Call Transfer by selecting it.

Semi-Attend
ed Transfer

Enable Semi-Attended Transfer by selecting it

Enable Auto
Handdown

The phone will hang up and return to the idle automatically at
hands-free mode

Auto
Handdown
Time

Specify Auto Hand down Time, the phone will hang up and
return to the idle automatically after Auto Hand down Time at
hands-free mode, and play dial tone Auto Hand down Time at
handset mode

Enable Auto
Redial

Enable Auto Redial by selecting it, then the phone reminds
whether redial, when the callee is busy or rejects

Auto Redial
interval

Specify the Auto Redial interval,

Auto Redial
Times

Specify the Auto Redial interval

Auto
Headset

Enable the function and put on the headset, when there has a
incoming call ,you can press the answer key or line key to
answer the call through the headset ,and it‟s the same if enable
auto answer function

Enable
Intercom

Enable Intercom Mode by selecting it

Enable
Intercom
Mute

Enable mute mode during the intercom call

Enable
Intercom
Tone

If the incoming call is intercom call, the phone plays the
intercom tone

Enable
Intercom
Barge

Enable Intercom Barge by selecting it, the phone auto answers
the intercom call during a call. If the current call is intercom
call, the phone will reject the second intercom call

P2P IP Prefix

Set Prefix in peer to peer IP call. For example: what you want
to dial is 192.168.1.119, If you define P2P IP Prefix as

192.168.1., you dial only #119 to reach 192.168.1.119. Default

is “.”. If there is no “.” Set, it means to disable dialing IP.

Turn Off
Power Light

Enable Turn Off Power Light by selecting it

Emergency
Call Number

Specify the Emergency Call Number. Despite the keyboard is
locked ,you can dial the emergency call number

Enable
Password
Dial

Enable Password Dial by selecting it, When number entered is
beginning with the password prefix, the following N numbers
after the password prefix will be hidden as *, N stand for the
value which you enter in the Password Length field. For
example: you set the password prefix is 3, enter

the Password Length is 2, then you enter the number 34567, it
will display 3**67 on the phone

Password
Dial Prefix

Specify the prefix of the password call number

Password
Length

Specify the Password length

Enable Call
Waiting

Enable Call Waiting by selecting it. then the phone reminds
whether redial, when the caller is busy or rejects . if it‟s ok and
the phone finds out that the caller is idle by sip message, it will
reminds whether redial

Enable
3-way
Conference

Enable 3-way conference by selecting it

Accept Any

Call

If select it, the phone will accept the call even if the called
number is IP .

Enable Call
Completion

Enable Call Completion by selecting it, If the callee is busy, the
sip server will inspect the callee status at intervals. If the callee
is idle, the server will send notify message to inform the caller
whether redial.

Enable
Pre-dial

Disable this feature, in standby interface next number, will
realize the number rules ”send out over the time”;Enable the
feature ,then the number will not be send out over the time.

Enable Silent
Mode

Enable Silent Mode by selecting it, the phone light will red
blink to remind that there is a missed call instead of playing
ring tone

Hide DTMF

Specify the hide DTMF mode

Ring From
Headset

Enable Ring From Handset by selecting it, the phone plays
ring tone from handset

DND Return
Code

Specify DND Return code

Busy Return
Code

Specify Busy Return Code

Reject
Return Code

Specify Reject Return Code

Active URI
Limit IP

Specify the server IP that remote control phone for
corresponding operation

Push XML
Server

Specify the Push XML Server, when phone receives request, it
will determine whether to display corresponding content on the
phone which sent by the specified server or not.

Enable Call
Waiting Tone

Turn off this feature, you will not hear issued a " beep" sound
with more calls

Enable Call
History

You can see the phone call history
Enable Multi

Enable multi-line,10 road calls can be enabled, otherwise make

Line

at most two road calls

Enable
Default Line

When enabled, a sip line becomes the default line; disabled,
the phone will begin in accordance with the line from the SIP1
sequential search, select an available line for calls.

Enable Auto
Switch Line

Prerequisite: Enable Default Line
ON: If sip1 registration fails, it will automatically find the first
available line down
Off: If sip1 registration fails, does not automatically find the
first available down the line, using the default line
If default Line is disabled, automatic line switch function does
not take effect

Allow IP
Call

Enter the remote IP to make a call

Play Talking
DTMF Tone

Press the number keys in a call, you can hear the sound of the
end

Play Dialing
tone

Press the number keys in dialing, you can hear the sound of the
end

Block Out
Settings

Block out

Set Add/Delete Limit List. Please input the prefix of those
phone numbers which you forbid the phone to dial out. For
example, if you want to forbid those phones of 001 as prefix to
be dialed out, you need input 001 in the blank of limit list, and
then you cannot dial out any phone number whose prefix is

001.

X and are wildcard x means matching any single digit. For
example, 4xxx expresses any number with prefix 4 which
length is 4 will be forbidden to dialed out mea ns matching any
arbitrary number digit. For example, 6 expresses any number
with prefix 6 will be forbidden to dialed out.

Notice: Black List and Limit List can record at most10 items respectively.

8.3.4.3 DIAL PLAN

This system supports 4 dial modes:

1) End with “#”: dial your desired number, and then press #.

2) Fixed Length: the phone will intersect the number according to your specified
length.

3) Time Out: After you stop dialing and waiting time out, system will send the
number collected.

4) Press # to Do Blind Transfer: input the number you want to transfer to then
press“#” you can transfer the current call to the number.

5) Blind Transfer on OnHook : input the number you want to transfer to then

hang up handle or press speaker, you can transfer the current call to the number.

6) Attend Transfer on OnHook: hang up handle or press speaker you can realize
the blind transfer function.

7) Press DSS Key to Do Blind Transfer: press dss-key, the phone will transfer
the current call to the third party

8) User defined: you can customize digital map rules to make dialing more
flexible. It is realized by defining the prefix of phone number and number length
of dialing.
In order to keep some users' secondary dialing manner when dialing the external
line with PBX, phone can be added a special rule to realize it. so user can dial a
number as external line prefix and get the secondary dial tone to keep dial the
external number. After finishing dialing, phone will send the prefix and external
number totally to the server.
For example, there is a rule 9, xxxxxxxx in the digital map table. After dialing 9,
phone will send the secondary dial tone, user may keep going dialing. After
finished, phone will call the number which starts with 9; actually the number
sent out is 9-digit with 9.

DIAL PLAN Configuration

Field name

explanation

Basic Setting

Press "#" to Send

Set Enable/Disable the phone ended with “#” dial.

Dial Fixed Length

Specify the Fixed Length of phone ending with.

Send after （3-30）

Set the timeout of the last dial digit. The call will be
sent after timeout.

seconds

Press # to Do Blind
Transfer

Enable Blind Transfer On Hook, when executing Blind
Transfer End with #, press # after inputting the number
that you want to transfer, the phone will transfer the
current call to the third party

Blind Transfer on
OnHook

Enable Blind Transfer on On Hook, when executing
Blind Transfer, hang up after inputting the number that
you want to transfer, the phone will transfer the current
call to the third party

Attend Transfer on
OnHook

Enable Attend Transfer on On Hook, when executing
Attended Transfer, hang up after the third party
answers, the phone will transfer the current call to the
third party

Press DSS Key to
Do Blind Transfer

Enable press DSS Key to Do Blind Transfer, when
executing Blind Transfer, press dss-key, the phone
will transfer the current call to the third party

Below is user-defined digital map rule:
[] Specifies a range that will match digit. May be a range, a list of ranges
separated by commas, or a list of digits.
* Match any single digit that is dialed.
. Match any arbitrary number of digits including none.
Tn Indicates an additional time out period before digits are sent of n seconds
in length. n is mandatory and can have a value of 0 to 9 seconds. Tn must be
the last 2 characters of a dial plan. If Tn is not specified it is assumed to be
T0 by default on all dial plans.

Cause extensions 1000-8999 to be dialed immediately
Cause 8 digit numbers started with 9 to be dialed immediately
Cause 911 to be dialed immediately after it is entered.
Cause 99 to be dialed after 4 seconds.
Cause any number started with 9911 to be dialed 4 seconds after dialing
ceases.

Notice: Dial plan can realised at speaker,pick handle or headset mode. End

with “#”, Fixed Length, Time out and Digital Map Table can be used

simultaneously, System will stop dialing and send number according

to your set rules.

8.3.4.4 CONTACT

You can input the name, phone number and select ring type for each name here.

CONTACT

Field name

explanation

Phonebook Table

Name

Shows the name corresponding to the phone number

Number

Shows the phone number

Ring Type

Shows the ring type of the incoming call.

Group

Shows the group of the contact

Notice: the maximum capability of the phonebook is 500 items, you can select
many or a contact to add to group and add to blacklist, and delete many or a
contact, and delete all contacts.

Add Contact List

Name

Specify the name corresponding to the phone
number

Office Number

Specify the office number

Mobile Number

Specify the mobile number

Other Number

Specify the other number

Ring Type

Specify the ring type for the phone number

Line

Specify the sip line for the each number

Group setting

Select the group from the unselected group to
selected list for the contact; you can select many
groups for the contact.

Notice: the add button for adding a new contact, the modify button for
modifying the added contact, the clear all button for clear all input information
of the contact

Group Option

Group

Select the added groups, then modify or delete and
so on

Name

Input the name of the group, then click the add
button, you can add a new group.

Ring Type

Specify the ring type for the group as adding a new
group

Import Contact List

Select File

Click the browse button to select the phonebook file
that you want to import, than click update button,
the phonebook file selected will be added to the
phone.

Export Contact File

Export XML

Click export xml button to export phonebook file of
xml model

Export CSV

Click export xml button to export phonebook file of
csv model

Export VCF

Click export xml button to export phonebook file of

vcf model

Blacklist Settings

Type

Select the blacklist type, you can select number or
prefix of number

Value

Input number or prefix of number

Line

Select the sip line

Notice: the add button for adding a new blacklist, the delete button for
deleting one item, the delete all button for deleting all items.
If user does not want to answer some phone calls, add these phone numbers to
the Black List, and these calls will be rejected.x and are wildcard x means
matching any single digit. for example, 4xxx expresses any number with

prefix 4 which length is 4 will be forbidden to be responded。
DOT (.) means matching any arbitrary number digit. for example, 6. expresses
any number with prefix 6 will be forbidden to be responded。
If user wants to allow a number or a series of number incoming, he may add

the number(s) to the list as the white list rule. the configuration rule is

-number, for example, -123456, or -1234xx

Means any incoming number is forbidden except for 4119
Note: End with DOT (.) when set up the white list

8.3.4.5 REMOTE CONTACT

You need to match a XML Phonebook address and you can directly access to the
corresponding remote phonebook on the phone.

Remote Phonebook
Setting

Phonebook Name

Custom the phonebook name displayed on the phone

Server URL

Specify the server url of the remote phonebook

Sip Line

Specify the sip line for the remote phonebook

Authentication

Specify the authentication mode for remote
phonebook

Username/password

Input the authentication username and password

（Note: remote book support the modes as HTTP ,FTP ,TFTP）

LDAP Settings

Display Title

LDAP phonebook name

Version

LDAP version

Server Address

LDAP server address

Server port

LDAP server port

Authentication

There are four options: NONE,
DIGEST-MD5,CRAM-MD5,Simple

Line

contacts call using the selected line

Username

Enter username

Password

Enter password

Search Base

query the root directory access to LDAP

Enable Calling Search

the phone search LDAP server in the outgoing /
incoming call . If a contact is searched, its name will
be displayed on the screen

Telephone

Contacts‟ telephone

Mobile

Contacts‟ mobile phone

Other

Contacts‟ other information

Display Name

Contacts‟ displayname

8.3.4.6 WEB DIAL

You can make a call through the WEB DIAL, enter the Dial Num then press Dial, if
you want to finish the talk, press Hang-up.

8.3.4.7 MCAST

Use the multicast function to send notice to every member of the multicast is
simple and easy. By setting the multicast key on your phone, you can send
multicast RTP flow to the pre-configured multicast address. By listening
multicast address is configured on the phone, listen and play the multicast
address to send the RTP stream.

Send multicast setting

On the phone web page,function key-function key,set a function key，as shown

Value format IP:Port，the IP address of multicast is range from 224.0.0.0 to

239.255.255.255,port is greater than 1024
If multicast codec is G722, the LCD screen will displays "HD", which means the
phone is sending high-definition voice stream
Operate steps:

1、 When the phone is idle, press multicast key
Multicast RTP stream is sended to pre-configured multicast address (IP:

Port)。The phone which listens to multicast address in the local network can

receive the RTP stream. Multicast functionkey LED lights yellow.
LCD screen displays the following:

2、 Press the hold softkey to hold the current multicast session
3、 Press the end softkey again or multicast functionkey, multicast session can

be stopped
Notice: RTP stream is one side, that is from a sender to a receiver. when the
phone initiates a multicast RTP session in a call,the current call is on hold.

Receive multicast setting

You can set up the phone monitoring 10 different multicast addresses to receive
these multicast RTP stream.
You have two method to receive RTP stream of multicast that can be set up
through the web page: Enable priorities of normal calls and Enable page
Priority:
Enable priorities of normal call by select it, if the incoming RTP stream
priority of multicast lower than the priority of current for normal calls, the
phone will ignore the RTP stream of multicast. If the incoming RTP stream
priority of multicast higher than the priority of current for normal calls,
the phone will receive the RTP stream of multicast, and hold the current call.
Disabled priorities of normal call by select disable, the phone will
ignore all local network RTP stream of multicast.
Options as follows:
1-10:the priority defined for normal calls,1 the highest level,10 the lowest level
Disabled: Ignore all RTP stream of multicast

Enable Page Priority
Page priority determines the phone how to handle the newly received multicast
RTP stream when in a multicast session. Enabled page priority, the phone will
automatically ignore the low priority multicast RTP stream and receive the high
priority multicast RTP stream and hold the current multicast session; If not
enabled, the phone will automatically ignore all incoming multicast RTP stream.
Web page is set as follows:

Now multicast ss has higher priority than multicast ee, the highest priority is for
normal calls
Notice: When a multicast session begins, multicast sender and receiver will beep

8.3.4.8 TONE

You can select the desired tone standard, also can customize the settings

8.3.4.9 Action URL

Specify the Action URL that Record the operation of phone, send these
corresponding information to server, url:http://InternalServer
/FileName.xml?(Internal Server is server ip, FileName is name of xml that
contains the action message )

8.3.5 Function Key

8.3.5.1 Function Key

Field name

explanation

Contrast

Set contrast of screen

Enable Backlight

Set enable/disable backlight

Backlight Time

You don‟t operate the phone more than a backlight

timeout setting, the backlight will turn off
automatically

Line Key Settings

Line: select Auto, SIP1, SIP2, SIP3, SIP4, or IAX2 in function key type.

After you set it, you pick up handset or hands-free, press this function
key, and then you can use the corresponding SIP line.

Function Key Settings

key

Show the function key's serial number

Type

Memory Key: settings can be stored in key storage
for each number, the standby or off-hook, select
the function keys on the keyboard can call this
number.
Line, set the dial mode (Auto, SIP1, SIP2, SIP3,
SIP4, IAX2).Key Key Event functions, monitor
state.

DTMF：In the call, send DTMF
URL: You can input remote book url

Value

Set the type parameter values.

Line

Choose which lines to use this feature.

Subtype

Select the function parameters Key Event and
Memory Event.

Pickup Number

The value of SubType is the number to BLF or

Presence.

NOTICE：

 memory keys can be configured through the following:

Speed Dial function, through the configuration of the key corresponding to
the number of ways as shown below.

User can press the F1 key to allocate this number by line1 line.
Intercom function, you can press this key in standby to automatically

answer the call and make each other.

User can be configured in accordance with push to talk function the way:
4116 was the other number; Then press the standby button and make it
automatically answer the call 4116.



key can be configured through the following events:

For example:

8.3.5.2 EXT KEY

EXT KEY has the same usage with the Function key. “In” port connects the

phone, “Out” port connects the next one, if there is only, you don‟t need for
power supply, if there are more than one, you need supply 5V power for the first
one, and use RJ-45 direct connector.

8.3.5.3 SOFTKEY

You can configure different functions in different screens for every softkey.

8.3.6 Maintenance

8.3.6.1 Auto Provision

Fanvil endpoint supports PnP and DHCP and Phone Flash to obtain the
parameters. The PnP and DHCP and Phone Flash are all deployed, endpoint will
go by the following process to try to obtain the server address and other
parameters, when it boots up:
DHCP option  

Auto Provision

Field name

explanation

Auto Update
Setting

Current Config
Version

Show the current config file‟s version. If the version
of the configuration downloaded is higher than the
version of the running configurations, the auto
provision would upgrade, or stop here. If the endpoints
confirm the configuration by Digest method, the
endpoints wouldn‟t upgrade configuration unless the
configuration in the server is different with the
running configuration.

Common Config
Version

Show the common config file‟s version. If the
configuration downloaded and the running

configurations are the same, the auto provision would
stop here. If the endpoints confirm the configuration
by Digest method, the endpoints wouldn‟t upgrade
configuration unless the configuration in the server is
different with the running configuration.

CPE Serial Number

Show CPE Serial Number

User

Specify FTP/HTTP/HTTPS server Username. System
will use anonymous if username keep blank.

Password

Specify FTP/HTTP/HTTPS server Password.

Config Encrypt Key

Input the Encrypt Key, if the configuration file is
encrypted.

Common Config
Encrypt Key

Input the Common Encrypt Key, if the Common
Configuration file is encrypted

Save Autoprovision
Information

Save the username and password authentication
message of http/https/ftp and input ID message in the
phone until the URL in the server changes

DHCP Option
Setting

DHCP Option
Setting

Specify DHCP Option. DHCP option supports DHCP
custom option and DHCP option 66 and DHCP option
43 to obtain the parameters. You could choose one

method among them，the default is DHCP option
disable.

Custom DHCP
Option

A valid Custom DHCP Option is from 128 to 254. The
Custom DHCP Option must be in accordance with the
one defined in the DHCP server.

Plug and Play

Enable PnP

Enable PnP by selecting it, than the phone will send
SIP SUBSCRIBE messages to a multicast address
when it boots up. Any SIP server understanding that
message will reply with a SIP NOTIFY message
containing the Auto Provisioning Server URL where
the phones can request their configuration.

PnP Server

Specify the PnP Server

PnP Port

Specify the PnP Server

PnP Transport

Specify the PnP Transfer protocol

PnP Interval

Specify the Interval time, unit is hour

Phone Flash

Server Address

Set FTP/TFTP/HTTP/HTTPS server IP address for
auto update. The address can be IP address or Domain
name with subdirectory.

Config File Name

Set configuration file‟s name which need to update.

System will use MAC as config file name if config file

name keep blank. For example, 000102030405.。

Protocol Type
Specify the Protocol type FTP、TFTP or HTTP.
Update Interval

Specify update interval time, unit is hour.

Update Mode

Different update modes:

1. Disable: means no update

2. Update after reboot: means update after reboot.

3. Update at time interval: means periodic update.

TR069 Settings

Enable TR069

Enable TR069 by selecting it

ACS Server Type

Specify the ACS Server Type

ACS Server URL

Specify the ACS Server URL

ACS User

Specify ACS User

ACS Password

Specify ACS Password

Periodix Interval

It will check every 6 minutes

TR069 Auto Login

Enable TR069 Auto Login by selecting it

"Inform" Sending
Period

Specify the “inform” Sending Period, unit is second

8.3.6.2 Syslog Config

Syslog is a protocol which is used to record the log messages with client/server
mechanism. Syslog server receives the messages from clients, and classifies
them based on priority and type. Then these messages will be written into log by
some rules which administrator can configure. This is a better way for log
management.
8 levels in debug information:
Level 0---emergency: This is highest default debug info level. You system cannot
work.
Level 1---alert: Your system has deadly problem.
Level 2---critical: Your system has serious problem.
Level 3---error: The error will affect your system working.
Level 4---warning: There are some potential dangers. But your system can work.
Level 5---notice: Your system works well in special condition, but you need to
check its working environment and parameter.
Level 6---info: the daily debugging info.
Level 7---debug: the lowest debug info Professional debugging info from R&D
person.
At present, the lowest level of debug information is info; debug level only can be
displayed on telnet.

Syslog Configuration

Field name

explanation

Syslog Setting

Server Address

Set Syslog server IP address.

Server Port

Set Syslog server port.

MGR Log Level

Set the level of MGR log.

SIP Log Level

Set the level of SIP log.

Enable Syslog

Select it or not to enable or disable syslog.

Watch Dog

Enable Watch Dog

Phone will automatically restart in case of failure

Web Capture

Start

Click the start button when you need capture the WAN
packet stream of the phone, then open or save the file
as the interface

Stop

Click the end button to stop capturing the packet
stream

Port mirror
Setting

When Port mirror Setting is enabled, device under the
LAN port can obtain the data from WAN port

8.3.6.3 Config

Config Setting

Field name

explanation

Save Configuration

You can save all changes of configurations. Click the
Save button, all changes of configuration will be
saved, and be effective immediately.

Backup
Configuration

Right clicks on “Right click here…” and select “Save

Target As config File(.txt)” then you will save the
config file in .txt format, or select “Save Target As
config File(.xml)” then you will save the config file
in .xml format

Clear Configuration

User can restore factory default configuration and
reboot the phone.
If you login as Admin, the phone will reset all
configurations and restore factory default; if you login
as Guest, the phone will reset all configurations except
for VoIP accounts (SIP1-4 and IAX2) and version
number.

8.3.6.4 Update

You can update your configuration with your config file in this web page.

Update

Field name

explanation

Web Update

Web Update

Click the browse button, find out the config file saved
before or provided by manufacturer, download it to the

phone directly, press “Update” to save. You can also

update downloaded update file, logo picture, ring,
mmiset file by web.

FTP Update

Server Address

Set the FTP/TFTP server address for
download/upload. The address can be IP address or
Domain name with subdirectory.

User

Set the FTP server Username for download/upload.

Password

Set the FTP server password for download/upload.

File name

Set the name of update file or config file. The default
name is the MAC of the phone, such as

000102030405.

Notice: You can modify the exported config file. And you can also download
config file which includes several modules that need to be imported. For
example, you can download a config file just keep with SIP module. After
reboot, other modules of system still use previous setting and are not lost.

Type

Action type that system want to execute：

1. Application update: download system update file

2. Config file export: Upload the config file to
FTP/TFTP server, name and save it.

3. Config fie import: Download the config file to
phone from FTP/TFTP server. The configuration will
be effective after the phone is reset.

4. Phone book export (.vcf, .csv, .xml): Upload the
phonebook file to FTP/TFTP server, name and save it.

5. PhoneBook import (.vcf, .csv, .xml): Download the
phonebook file to phone from FTP/TFTP server.

Protocol

Select FTP/TFTP server

Update Logo File

Select File

Specify the URL of the logo file

Delete Logo File

Select File

Select the logo that you want to delete

Logo File

Logo File

Show the logo file

8.3.6.5 ACCESS

You can add or delete user account, and change the authority of each user
account in this web page.

Access Configuration

Field name

explanation

Keyboard Password

Set the password for entering the setting menu of the

phone by the phone„s key board. The password is

digit.

This table shows the current user existed.

User

Set account user name.

User Level

Set user level, Root user has the right to modify
configuration, General can only read.

Password

Set the password.

Confirm

Confirm the password.

Select the account and click the Modify to modify the selected account, and
click the Delete to delete the selected account.
General user only can add the user whose level is General.

8.3.6.6 REBOOT

If you modified some configurations which need the phone‟s reboot to be

effective, you need click the Reboot, then the phone will reboot immediately.
Notice: Before reboot, you need confirm that you have saved all configurations.

8.3.7 Security

8.3.7.1 Web Filter

WEB Filter

User could make some device own IP, which is pre-specified, access to the
MMI of the phone to config and manage the phone.

Field name

explanation

Web Filter Table Settings:

Add or delete the IP address segments that access to the phone.

Set initial IP address in the Start IP column, Set end IP address in the End IP
column, and click Add to add this IP segment. You can also click Delete to
delete the selected IP segment.

Web Filter setting

Select it or not to enable or disable Web Filter. Click

Apply to make it effective.

Notice: Do not set your visiting IP outside the Web filter range, otherwise,

you cannot logon through the web.

8.3.7.2 Firewall

Firewall Configuration

In this web interface, you can set up firewall to prevent unauthorized Internet
users from accessing private networks connected to the Internet (input rule),
or prevent unauthorized private network devices from accessing the Internet
(output rule).
Firewall supports two types of rules: input access rule and output access rule.
Each type supports at most 10 items.
Through this web page, you could set up and enable/disable firewall with
input/output rules. System could prevent unauthorized access, or access other
networks set in rules for security. Firewall, is also called access list, is a
simple implementation of a Cisco-like access list (firewall). It supports two

access lists: one for filtering input packets, and the other for filtering output
packets. Each kind of list could be added 10 items.
We will give you an instance for your reference.

Field name

explanation

Enable Input Rules

Select it to Enable Input Rules

Enable Output
Rules

Select it to Enable Output Rules

Input / Output

Specify current adding rule by selecting input rule or
output rule.

Deny/Permit

Specify current adding rule by selecting Deny rule or
Permit rule.

Protocol

Filter protocol type. You can select TCP, UDP, ICMP,
or IP.

Port Range

Set the filter Port range

Src Address

Set source address. It can be single IP address,
network address, complete address 0.0.0.0, or network
address similar to *.*.*.0

Des Address

Set the destination address. It can be IP address,
network address, complete address 0.0.0.0, or network
address similar to *.*.*.*

Src Mask

Set the source address‟ mask. For example,

255.255.255.255 means just point to one host;

255.255.255.0 means point to a network which
network ID is C type.

Dest Mask

Set the destination address‟ mask. For example,

255.255.255.255 means just point to one host;

255.255.255.0 means point to a network which
network ID is C type.

Click the Add button if you want to add a new output rule.
Then enable out access, and click the Apply button.
So when devices execute to ping 192.168.1.118, system will deny the request
to send icmp request to 192.168.1.118 for the out access rule. But if devices
ping other devices which network ID is 192.168.1.0, it will be normal.

Click the Delete button to delete the selected rule.

Click the Add button if you want to add a new output rule.
Then enable out access, and click the Apply button.
So when devices execute to ping 192.168.1.118, system will deny the request
to send icmp request to 192.168.1.118 for the out access rule. But if devices
ping other devices which network ID is 192.168.1.0, it will be normal.

Click the Delete button to delete the selected rule.

8.3.7.3 VPN

This web page provides us a safe connect mode by which we can make remote
access to enterprise inner network from public network. That is to say, you can
set it to connect public networks in different areas into inner network via a
special tunnel.

VPN Configuration

Field name

explanation

VPN IP

Shows the current VPN IP address

Select L2TP. You can choose only one for current state. After you select it,
you‟d better save configuration and reboot your phone.

Enable VPN
Select it or not to enable or disable VPN；
VPN Server Addr

Set VPN L2TP Server IP address

VPN User Name

Set User Name access to VPN L2TP Server

VPN Password

Set Password access to VPN L2TP Server

8.3.7.4 Security

Security

Field name

explanation

Update Security
File

Select Security File

Select the security file you want to update, then click
Update button to update

Delete Security File

Select Security File

Select the security file you want to delete, then click
Delete button to update

SIP TLS File

Show SIP TLS authentication certification file

HTTPS File

Show HTTPS authentication certification file

Open VPN Files

Show Open VPN File authentication certification file

8.3.8 Logout

Click Logout，and you will exit web page. If you want to enter it next time, you
need input user name and password again.

9 Appendix

9.1 Specification

9.1.1 Hardware

Item

DPH-400G/DPH-400GE

Adapter
(Input / Output)

Input: 100-240V
Output: 5V 1A

port

WAN

10/100Base- T RJ-45 1 PORT

LAN

10/100Base- T RJ-45 1 PORT

EXT

RJ11 1 PORT

Power
Consumption

Idle: 2.5W/Active: 2.8W

LCD Size

128x64

53.5 x 70mm

Operation
Temperature

0～40℃

Relative Humidity
10～65%

CPU

Broadcom

SDRAM

64MB

Flash

128MB

Dimension(L x W x
H)

295×295×175mm
Weight

1.5kg

9.1.2 Voice features

 SIP supports 5 SIP servers
 Support SIP 2.0 (RFC3261) and correlative RFCs

 Codec: G.711A/u, G.722, G.723.1,G.726-32,G.729.AB,ILBC, AMR,

AMR-WB

 Echo cancellation: G.168 Compliance in LEC, additional acoustic echo

cancellation(AEC) can reach 96ms max filter length in hands-free mode

 Support Voice Gain Setting, VAD, CNG
 Support full duplex hands-free
 Support multi line/HD Voice

- 95 -

 SIP support SIP domain, SIP authentication(none basic, MD5), DNS name

of server, Peer to Peer/ IP call

 Automatically select calling line, if one line can‟t be connected, the phone

can automatically switch to other line to call.

 9 kinds of ring types and 3 user-defined music rings
 DTMF Relay: support SIP info, DTMF Relay, RFC2833

 SIP application: SIP Call forward/transfer（blind/attended）/hold/waiting/3

way talking/SMS/pickup /join call /redial /unredial/multi
line/intercom/BLF/presence/push to talk/auto redial/call return

 Call control features: Flexible dial map, hotline, empty calling No. reject

service, black list for reject authenticated call, white list, limit call, no disturb,
caller ID, CLIR(reject the anonymous call), CLIP(make a call with
anonymous), Dial without register.

 Support phonebook 500 records, Incoming calls / outgoing calls / missed

calls. Each supports 100 records.

 6 line keys defined as multi line with screen display or used as SIP line keys
 8 DSS keys
 Soft keys programmable, function keys programmable
 Code synchronization via IP PBX/IMS
 Support EXT DSS consoles with 5 max
 Support click to dial via web phone book
 Voice codec setting for each SIP line
 Support keypad lock, and emergency call during the keypad lock
 Customized lcd logo
 Ring play via headset or speaker setting
 Signal tone parameters customized
 Phonebook supports vcard standard
 12/24 hours time display
 Support daylight saving time
 Support path, group
 Support SIP Privacy
 Support SMS
 Support WMI
 Support Speed dial
 Support XML

9.1.3 Network features

 WAN: support bridge model
 Support PPPoE for xDSL
 Support VLAN (optional: voice vlan/ data vlan)
 Support VPN (L2TP) function

- 96 -

 Wan Port supports main DNS and secondary DNS server, can select

dynamically to get DNS in DHCP mode or statically set DNS address.

 Support DHCP client on WAN
 QoS with DiffServ
 Network tools in telnet server: including ping, telnet client
 Support Gigabit Ethernet

9.1.4 Maintenance and management

 Upgrade firmware through POST mode
 Web ,telnet and keypad management
 Management with different account right
 LCD and WEB configuration can be modified into requested language, and

support multi-language dynamically shifted

 Upgrade firmware through HTTP,HTTPS FTP or TFTP Telnet remote

management/ upload/download setting file

 Support Syslog
 Support Auto Provisioning (upgrade firmware or configuration file)

9.2 Digit-character map table

Keypad

Character

Keypad

Character

1 @

7 P Q R S p q r s

2 A B C a b c
8 T U V t u v

3 D E F d e f

9 W X Y Z w x y z

4 G H I g h i

*/.

5 J K L j k l

0

6 M N O m n o

#/=